1. Field of the Invention
This invention relates to an information recording and reproducing system and method and a distribution medium and in particular to an information recording and reproducing system and method and a distribution medium capable of providing a user with predetermined information from a plurality of types of information as required.
2. Description of the Related Art
A software distribution method is known wherein acoustic signals, etc., of a piece of music, for example, are encrypted and broadcasted or are recorded on a record medium and only the person buying a key is allowed to listen to the piece of music. Known as an encryption method is a method wherein, for example, an initial value of a random number sequence is given as a key signal and a bit string resulting from exclusive-ORing the random number sequence of “0” or “1” generated and a bit string of PCM of an acoustic signal is transmitted or is recorded on a record medium. This method makes it possible for only the person taking possession of a key signal to be able to reproduce the acoustic signal and for the person not taking possession of the key signal to be able to reproduce only noise.
On the other hand, a method of compressing an acoustic signal and broadcasting the compressed signal or recording the compressed signal on a record medium becomes pervasive and is used to record signals of audio, voice, etc., coded on a record medium such as a magneto-optic disc. Various techniques of highly efficient coding of signals of audio, voice, etc., are available; for example, the following coding techniques can be named: Band split coding (subband coding (SBC)) of an unblock frequency band splitting method of splitting audio signal, etc., on the time axis into frequency bands and coding without blocking and a block frequency band splitting method of transforming a signal on the time axis into a signal on the frequency axis (spectrum transform) and splitting the signal into frequency bands and coding for each band, so-called transform coding.
A highly efficient coding technique using the subband coding (SBC) and the transform coding in combination is also possible. In this case, for example, band splitting is executed according to the subband coding, then the signal for each band is spectrum-transformed into a signal on the frequency axis and coding is executed for each band provided by the spectrum transform.
For example, a QMF filter is available as a filter for executing the band splitting; it is described in 1976 R. E. Crochiere Digital coding of speech in subbands Bell Syst. Tech. J. Vol. 55, No. 8 1976.
A filter splitting method of an equal bandwidth is described in    ICASSP 83, BOSTON Polyphase Quadrature filters—A new subband coding technique    Joseph H. Rothweiler
Further, the spectrum transform includes, for example, spectrum transform wherein, for example, an input audio signal is blocked according to a predetermined unit time (frame) and discrete Fourier transform (DFT), discrete cosine transform (DCT), modified discrete cosine transform (MDCT), etc., are executed for each block, whereby the time axis is transformed into the frequency axis. The MDCT is described in    ICASSP 1987    Subband/Transform Coding    Using Filter Bank Designs Based on Time Domain Aliasing Cancellation    J. P. Princen A. B. Bradley Univ. of Surrey Royal Melbourne Inst. of Tech.
When the DFT or the DCT is used as a method of transforming a waveform signal into a spectrum, if transform is executed in a time block consisting of M samples, M independent real data pieces are provided. To decrease connection distortion between time blocks, normally the block is made to overlap both adjacent blocks each M1 samples. Thus, for (M–M1) samples, M real data pieces are quantized and coded in the DFT or the DCT on average.
In contrast, if the MDCT is used as a method of transforming into a spectrum, M independent real data pieces are provided from 2M samples made to overlap both adjacent times each M pieces. Thus, for M samples, M real data pieces are quantized and coded in the MDCT on average. In a decoder, waveform elements provided by executing inverse transform in each block from the code thus provided using the MDCT are added together while they are made to interfere with each other, whereby the waveform signal can be reconstructed.
Generally, the time block for transform is lengthened, whereby the spectrum frequency resolution is raised and energy concentrates on a specific spectrum component. Therefore, the block is made to overlap both adjacent blocks each a half, transform is executed in long block length, and moreover the MDCT is used wherein the number of spectrum signals provided does not increase with respect to the number of original time samples, whereby it is made possible to execute more efficient coding as compared with the case where the DFT or the DCT is used. The adjacent blocks are provided with sufficiently long overlap, whereby distortion between the blocks of waveform signal can also be decreased.
The quantization noise occurring band can be controlled by quantizing the signal split for each band through the filter or by the spectrum transform in such a manner, and the nature of the masking effect, etc., can be used to execute more highly efficient coding as auditory sense. If normalization is executed for each band before execution of quantization, for example, with the maximum value of the absolute values of signal components in the band, furthermore highly efficient coding can be executed.
For example, band splitting considering the auditory characteristic of a human being is executed as the frequency split width for quantizing each frequency component subjected into frequency band splitting. That is, audio signal may be split into bands (for example, 25 bands) with band width such that the band width widens as the band becomes higher generally called critical band. At this time, to code data for each band, the data is coded based on predetermined bit assignment for each band or adaptive bit allocation for each band. For example, when coefficient data provided by performing MDCT processing is coded according to bit allocation, the MDCT coefficient data for each band provided by performing MDCT processing for each block is coded according to the adaptive number of allocated bits. As bit allocation techniques, the techniques described in the following documents are known:    Adaptive Transform Coding of Speech Signals    R. Zelinski and P. Noll    IEEE Transactions of Acoustics, Speech, and, Signal Processing, vol.ASSP-25, No. 4, August 1977describe a technique of allocating bits based on the signal magnitude for each band. In this technique, a quantization noise spectrum becomes flat and noise energy reaches the minimum, but as the auditory sense, the masking effect is not used and thus the actual noise feel is not optimum feel.    ICASSP 1980    The critical band coder            —digital encoding of the perceptual requirements of the auditory system            M. A. Kransner MITdescribes a technique wherein auditory masking is used for obtaining signal-to-noise ratio required for each band and fixed bit allocation is executed. In this technique, however, to measure the characteristic on sine wave input, bit allocation is also fixed and thus the characteristic value does not become a very good value. To solve this problem, a highly efficient coder is proposed wherein all bits used for bit allocation are divided into those for a fixed bit allocation pattern predetermined for each small block and those for bit allocation depending on the signal magnitude in each block, the division ratio is made to depend on the signal related to an input signal, and the smoother the signal spectrum, the larger the division ratio to the bits for a fixed bit allocation pattern.
According to the method, if energy concentrates on specific spectrum as with sine wave input, many bits are allocated to the block containing the spectrum, whereby the whole signal-to-noise characteristic can be improved remarkably. Generally, the auditory sense of a human being is extremely sensitive to a signal having a steep spectrum component. Thus, improving the signal-to-noise characteristic by using such a method not only leads to improving the numeric value on measurement, but also is effective for improving the sound quality on the auditory sense.
In addition to the bit allocation method, a large number of methods are proposed. If the model for the auditory sense is made finer and the capability of a coder is enhanced, more highly efficient coding is enabled from the viewpoint of the auditory sense. In the methods, it is a common practice to find such a real bit allocation reference value to realize the signal-to-noise characteristic found by calculation as faithfully as possible and adopt the integer value approximating the real bit allocation reference value as the number of allocated bits.
The present applicant previously proposed a method of separating the tone component particularly important on the auditory sense, namely, the signal component with energy concentrating on a specific frequency periphery from a spectrum signal and coding the signal component aside from any other spectrum component, whereby it is made possible to code an audio signal, etc., efficiently at a high compression rate scarcely causing degradation on the auditory sense (Japanese Patent Application NO. Hei 7-500482).
To form an actual code string, first, quantization precision information and normalization coefficient information may be coded in a predetermined number of bits for each band where normalization and quantization are executed, next a normalized and quantized spectrum signal may be coded.    ISO/IEC 11172-3:1993 (E)describes a highly efficient coding technique set so that the number of bits representing quantization precision information varies from one band to another as standardization wherein the number of bits representing quantization precision information lessens as the band becomes higher.
A method of determining the quantization precision information from the normalization coefficient information, for example, in a decoder instead of directly coding the quantization precision information is also known. However, in this method, when a standard is set, the relationship between the quantization precision information and normalization coefficient information is determined, thus it is made impossible to introduce control of quantization precision based on a more advanced auditory sense model in the future. If there is a width of the compression rates realized, it becomes necessary to define the relationship between the normalization coefficient information and quantization precision information for each compression rate.
A more efficient coding method is also known wherein a quantized spectrum signal is coded using variable-length code, for example, described in    D. A. Huffman: A Method for Construction of Minimum Redundancy Codes, Proc. I.R.E., 40, p. 1098 (1952).
It is also possible to encrypt a signal coded as described above as with PCM signal and distribute the encrypted signal. In this case, the person not taking possession of the key signal cannot reproduce the original signal. A method of converting a PCM signal into a random signal and then coding the signal for compression rather than encrypting a coded bit string is also available. In this case, the person not taking possession of the key signal can reproduce only noise.
However, in the scrambling methods, if the signal is reproduced without the key or by a normal reproducer, noise results and the contents of the software cannot be understood. Thus, the methods cannot be used for applications wherein, for example, a disc on which music is recorded with comparatively low sound quality is distributed and the person listening to the music recorded on the disc purchases a key only to his or her favorite piece of music so that he or she can play back the piece of music with high sound quality, or he or she can listen to software before purchasing a new disc on which the software is recorded with high sound quality.
Hitherto, to encrypt a highly efficiently coded signal, it has been difficult not to degrade the compression efficiency while giving a significant code string for a normal reproducer. That is, as described above, when a code string provided by highly efficient coding is scrambled, if the code string is reproduced, noise occurs and in addition, the reproducer does not operate at all if the scrambled code string does not comply with the standard of the original highly efficient coding. In contrast, if a PCM signal is scrambled and then highly efficiently coded, as the information amount is cut using the nature of the auditory sense, it becomes difficult to descramble correctly because the signal resulting from scrambling the PCM signal cannot always be reproduced when the highly efficiently coded signal is decoded. Thus, a method capable of descrambling correctly needs to be selected as the compression method even if the efficiency is degraded.
In view of the actual circumstances, the applicant proposed a method of coding the components of all bands of audio signal as shown in FIG. 1 and formatting with frames as record units, then recording the result on a record medium as shown in FIG. 2 and a method of separating audio signal into low-band signals (Q(1) to Q(C)) and high-band signals (Q(C+1) to Q(B)) and encrypting only the high-band portions (R(Q(C+1)) to R(Q(B)) as shown in FIG. 3, thereby enabling the listener to understand the contents of the low-band signals recorded with sound quality with narrow reproduction band with no key and requesting the user to determine whether or not he or she is to take possession of the key required for decoding the high-band portions based on the listening result (Japanese Patent Application No. Hei 8-288542).
The applicant also proposed a method wherein an acoustic signal of commentary voice added to an acoustic signal of music, etc., is coded in a first coding method and a cancel signal for canceling the acoustic signal of commentary voice is coded by a second coding method with added processing of encryption, etc., as shown in FIG. 4, whereby the listener can play back the music signal with the commentary voice using a reproducer that can decode and reproduce only the code provided by the first coding method and can play back the pure music signal with the commentary voice canceled using a reproducer that can decode and reproduce the code provided by the first and second coding methods (Japanese Patent Application No. Hei 9-301093).
However, in the method of canceling the commentary voice by the cancel signal, for example, if it becomes necessary to insert commentary voice into a silent part of music data, quantization noise occurring for the commentary voice signal and that occurring for the commentary voice cancel signal do not completely cancel out and moreover a signal for masking noise (music) does not exist, thus noise is conspicuous in the silent section of music. Thus, to insert commentary voice, it is necessary to device a plan for skipping a silent part, etc.